| Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc | 
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc | 
| index 252ffb2a3e59737e125e10c605b137ec9d67aeca..9d134afbccc81a934c0146af9ffb116395dc1ac1 100644 | 
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc | 
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc | 
| @@ -695,6 +695,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) { | 
| size_t length = IP_PACKET_SIZE; | 
| uint8_t data_buffer[IP_PACKET_SIZE]; | 
| int64_t capture_time_ms; | 
| + | 
| if (!packet_history_.GetPacketAndSetSendTime(packet_id, min_resend_time, true, | 
| data_buffer, &length, | 
| &capture_time_ms)) { | 
| @@ -922,8 +923,8 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer, | 
| // TODO(sprang): Potentially too much overhead in IsRegistered()? | 
| bool using_transport_seq = rtp_header_extension_map_.IsRegistered( | 
| kRtpExtensionTransportSequenceNumber) && | 
| -                             transport_sequence_number_allocator_ && | 
| -                             !is_retransmit; | 
| +                             transport_sequence_number_allocator_; | 
| + | 
| PacketOptions options; | 
| if (using_transport_seq) { | 
| options.packet_id = | 
|  |